This invention is related to a coding/decoding method and a coding/decoding apparatus for coding a signal or decoding a coded signal. More specifically, it is related to a coding/decoding method and a coding/decoding apparatus which is suitable for obtaining a high quality decoded acoustic signal at a low bit rate.
Recently, a number of techniques for coding wide-band speech or an acoustic signal have been proposed for use in multi-media. These techniques often use the adaptive transform coding method, in which an acoustic signal in a time domain is transformed into one in a frequency domain and, using the spectral envelope thereof, the coding is accomplished by adaptably determining a bit allocation or the like on a frequency axis. This is because in the adaptive transform coding method speech quality is not affected by the input acoustic signal and the bit rate can be low due to the application of auditory masking effects. Such coding/decoding methods are disclosed in Japanese Patent Application laid open No. 3-184098, and "Transform Coding of Acoustic Signals Using Perceptual Noise Criteria: James D. Johnston: IEEE Journal on Selected Areas in Communications, Vol. 6, No. 2".
The above mentioned adaptive transform coding/decoding method will now be briefly explained with reference to FIG. 7 which is a block diagram of a system to which the adaptive transform coding/decoding method is applied. As shown in FIG. 7, the system 10' comprises a coding device 12' and a decoding device 14'.
The coding device 12' comprises a buffer 16 which receives a digital acoustic signal supplied from an analog-to digital (A/D) converter (not shown) and temporarily stores it in coding blocks each consisting of an acoustic signal of appropriate data length, a fast Fourier transform (FFT) section 18' connected to the buffer 16 for receiving each coding block from the buffer 16 and subjecting it to fast Fourier transformation, a spectral envelope calculation section 20 connected to the buffer 16 for producing a spectral envelope of the coding block received from the buffer 16, a spectral envelop coding section 22 which produces a spectral envelope code and a coded spectral envelope based on the spectral envelope produced by the spectral envelope calculation section 20, a transform coefficient normalization section 24 which receives transform coefficients produced by the FFT section 18' and the coded spectral envelope produced by the spectral envelop coding section 22 and produces normalized transform coefficients which are normalizations of the transform coefficients, a bit allocation calculation section 26 which receives the coded spectral envelope and calculates a bit allocation for quantizing the normalized transform coefficients, a transform coefficient quantization section 28 which quantizes the normalized transform coefficients based on the bit allocation calculated by the bit allocation calculation section 26, and a multiplexer 30 which outputs a digital transmission code obtained by multiplexing the quantized normalized transform coefficient code and spectral envelope code. The digital transmission code produced by the thus structured coding device 12' is stored in a storage medium such as an optical disk or transferred to the decoding device 14' via a communication line.
On the other hand, the decoding device 14' comprises a de-multiplexer 32 which de-multiplexes the digital transmission code received from a storage medium such as an optical disk or from the multiplexer 30 of the coding device 12' to obtain quantized normalized transform coefficients and a spectral envelope code, a spectral envelope decoding section 34 which receives the spectral envelope code and decodes it, a bit allocation calculation section 36 which calculates a bit allocation based on the spectral envelope produced by the spectral envelope decoding section 34, a transform coefficient inverse-quantization section 38 which inverse-quantizes the quantized normalized transform coefficient based on the bit allocation calculated by the bit allocation calculation section 36, a transform coefficient restore section 40 which restores the transform coefficients based on the spectral envelope produced by the spectral envelope decoding section 34, an inverse FFT section 42' which performs inverse fast Fourier transformation based on the transform coefficients restored by the transform coefficient restore section 40, and a buffer 44 which temporarily stores each signal (coding block) produced by the inverse FFT section 42'. The coding block temporarily stored in the buffer 44 is read out in an appropriate manner, whereby an acoustic signal 45 can be obtained.
It is well known that components having large power often concentrate in the low frequency band of the acoustic signal. The above mentioned adaptive transform coding method can be called a technique for obtaining codes of low distortion and high compression rate efficiently using unevenness of power in a frequency band.
However, it is also known that in the adaptive transform coding method, decreasing the bit rate in order to increase the compression rate increases signal degradation. This is because the adaptive quantization of the normalized transform coefficients often produces no bit for certain frequency bands, especially high frequency bands generally having low power.
In the conventional adaptive transform coding method, an attempt is made to solve this problem by setting that the normalized transform coefficient in a band having no allocated bit to 0 (zero) or to a random value. If the number of bands to which one of those techniques is applied is relatively small in the adaptive transform coding method, problems seldom occur. However, if the technique is applied to a number of successive bands at an extremely low bit rate, the sound quality is degraded to an extent noticeable to listeners. The degradation includes dropout of bands due to setting the normalized transform coefficient to 0 (zero) and occurrence of noise due to setting it to the random value. This disarranges the harmonic elements of the acoustic signal, which results in a serious problem.
In other words, the conventional adaptive transform coding method does not sufficiently deal with the degradation owing to the occurrence of a number of bands having no allocated bits and is therefore not sufficient for coding an acoustic signal at an extremely low bit rate.